1. Technical Field of the Invention
The present invention relates to decoding transmitted signals, and particularly to decoding transmitted signals based upon the state of the corresponding physical layer headers.
2. Description of the Related Art
With the introduction of new services or applications over packet data systems in a mobile communication network, for example, real time (RT) services such as VoIP, there will be a large variety of quality of service (QoS) demands on the network. Certain users, for example, those utilizing real time voice applications will have a very high demand for the availability of transmission resources, whereas users, for example, who transmit short messages or electronic mail, will be satisfied with a lower availability of transmission resources.
For example, in the well known Universal Mobile Telecommunications System (UMTS), there are four proposed QoS classes: the conversational class; streaming class; interactive class; and background class. The main distinguishing factor between these classes is the sensitivity to traffic delay. Conversational class traffic is intended for traffic which is very delay sensitive while background class traffic is the most delay insensitive traffic class. Conversational and streaming classes are intended to be used to carry RT traffic flows and interactive and background classes are intended to be used to carry Internet applications (e.g., www, e-mail, telnet, FTP, etc.).
Real time services include sensitive time constraints over a reserved access channel. That is, delays in the transmission and/or reception of successive packets can have noticeable and undesirable QoS effects (e.g., on voice quality). These time constraints can be handled by always reserving access time at predetermined intervals during a communication with high QoS demands. In this way, a real time service communication can proceed uninterrupted since it will be allocated communication resources regardless of whether or not any packets will be sent. That is, for example, silent periods will occur in a real time voice communication, and in order to conserve battery resources, the silent periods need not be transmitted, or the transmission power can be considerably reduced.
Silent periods can be detected in a voice activity detector (VAD) device. During silent periods, a silence descriptor (SID) signal is sent to the receiver. The receiver generates comfort noise in order to closely mimic the naturally occurring background noise so that the receiving user perceives that the communication path between the transmitter and the receiver is still open and operable. In addition to the SID, an indication is sent to the transmitter that there is no voice activity detected and the transmitter can reduce its transmitter output power or set it to zero for that connection. This technique is called discontinuous transmission (DTX). With DTX enabled, interference is decreased in the system, since transmitters will only emit output power when there is information to be transmitted (e.g., when voice activity is detected or when SIDs are transmitted).
Resources are allocated for the real time services users regardless of whether or not packets are sent from the transmitter. However, it would be advantageous if these silent periods could be used in a more efficient way by allowing other applications to use the allocated resources during the silent periods without lowering the QoS of real time service.
In connection with the development of third generation mobile communication systems, new wireless multimedia and data applications are being designed and introduced. To support these new applications, improved data transmission technologies are also being developed. One such technology is Enhanced Data rates for Global Evolution (EDGE), which uses a more efficient radio-modulation technology that is optimized for data communications and that can be implemented on existing GSM and IS-136 systems. When used in connection with General Packet Radio Service (GPRS), a packet-switched technology that delivers speeds of up to 115 kilobits per second (kbit/s), EDGE technology can increase end user data rates up to 384 kbit/s, and potentially higher in high quality radio environments.
In connection with the development of EDGE and other technologies for supporting higher data rates, a number of techniques for multiplexing different users on the same set of resources have been developed. For example, in the packet-switched mode of EDGE technology (i.e., Enhanced GPRS (EGPRS)), the capability exists to multiplex different users on the same time slot. In this mode, packet data is transferred via a wireless communication link using 20 millisecond (ms) radio blocks. Each radio block is transferred to or from a particular user as a sequence of four consecutive bursts on a time slot that is assigned to the user. Subsequently, the time slot can be assigned to another user for the transmission of four bursts to or from that other user or can be again assigned to the same user for the transmission of an additional four bursts.
When transmitting information having different formats on the same channel, a receiver needs to know the current format of a transmitted signal frame in order to perform successful decoding of the information therein. For instance, the physical layer headers and/or stealing bits associated with signals transmitted in GSM systems inform the receiver whether transmitted information is speech or the fast associated control channel (FACCH). In a GSM/EDGE Radio Access Network (GERAN), however, information other than speech and FACCH may be transmitted. This is possible because many voice coders, such as the Adaptive Multi-Rate (AMR) coder, utilize DTX, and the silence periods can then be used to transmit best effort data (such as interactive and background classes).
Within the receiving side of an AMR communications link, there resides a receiver DTX handler. The receiver DTX handler indicates to the speech decoder whether received signals are speech or comfort noise. The receiver DTX handler transitions, between two possible states. In a first state, the SPEECH state, the receiver DTX handler is configured to deliver speech signals to the AMR vocoder. During a speech silence period the receiver DTX handler enters a second state, the COMFORT_NOISE state, and generates comfort noise as is known in the art. The receiver DTX handler searches for various AMR identification markers, such as SID_FIRST (which marks the beginning of a DTX period) and SID_UPDATE, the reception of which causes the receiver to transition to the COMFORT_NOISE state from the SPEECH state. When a speech frame is correctly decoded, the start of a talk spurt is implicitly indicated and the receiver DTX handler transitions to the SPEECH state where comfort noise generation is suspended.
More recent GERAN systems require the capability to decode additional formats at the receiver during speech silence periods. In addition to speech and FACCH, additional formats include the Packet Associated Control Channel (PACCH) and various types of data, such as Modulation & Coding Schemes (MCS) 1-9. Prior attempts to handle additional formats and/or the capability of decoding additional formats include extending the length of the physical layer header in order to accommodate a greater number of different format indications therein. This approach, however, increases overhead which in turn decreases communication performance. This is especially critical when transmitting overhead-sensitive information, such as speech information. Further, changing the size of the physical layer header may lead to changing existing header coding assignments, thereby undesirably resulting in changing standardization, implementation and verification.
Another prior attempt to handle additional information formats include letting the receiver be state dependent, and having the receiver to perform an exhaustive search for the correct format among the ones allowed in each state. In this approach, the receiver decodes the format and uses CRC to verify the successfully decoded format. The approach, however, results in a substantial increase in decoding complexity since the receiver's decoder is potentially forced to decode each block of information several times before proceeding.
What is needed, then, is a receiver that is capable of efficiently and effectively handling transmitted speech and other information.